+++ /dev/null
-/*
- * This file is part of DisOrder
- * Copyright (C) 2005-2008 Richard Kettlewell
- *
- * This program is free software: you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program. If not, see <http://www.gnu.org/licenses/>.
- */
-/** @file server/speaker-network.c
- * @brief Support for @ref BACKEND_NETWORK */
-
-#include "common.h"
-
-#include <unistd.h>
-#include <poll.h>
-#include <netdb.h>
-#include <gcrypt.h>
-#include <sys/socket.h>
-#include <sys/uio.h>
-#include <net/if.h>
-#include <ifaddrs.h>
-#include <errno.h>
-#include <netinet/in.h>
-
-#include "configuration.h"
-#include "syscalls.h"
-#include "log.h"
-#include "addr.h"
-#include "timeval.h"
-#include "rtp.h"
-#include "ifreq.h"
-#include "speaker-protocol.h"
-#include "speaker.h"
-
-/** @brief Network socket
- *
- * This is the file descriptor to write to for @ref BACKEND_NETWORK.
- */
-static int bfd = -1;
-
-/** @brief RTP timestamp
- *
- * This counts the number of samples played (NB not the number of frames
- * played).
- *
- * The timestamp in the packet header is only 32 bits wide. With 44100Hz
- * stereo, that only gives about half a day before wrapping, which is not
- * particularly convenient for certain debugging purposes. Therefore the
- * timestamp is maintained as a 64-bit integer, giving around six million years
- * before wrapping, and truncated to 32 bits when transmitting.
- */
-static uint64_t rtp_time;
-
-/** @brief RTP base timestamp
- *
- * This is the real time correspoding to an @ref rtp_time of 0. It is used
- * to recalculate the timestamp after idle periods.
- */
-static struct timeval rtp_time_0;
-
-/** @brief RTP packet sequence number */
-static uint16_t rtp_seq;
-
-/** @brief RTP SSRC */
-static uint32_t rtp_id;
-
-/** @brief Error counter */
-static int audio_errors;
-
-/** @brief Network backend initialization */
-static void network_init(void) {
- struct addrinfo *res, *sres;
- static const struct addrinfo pref = {
- .ai_flags = 0,
- .ai_family = PF_INET,
- .ai_socktype = SOCK_DGRAM,
- .ai_protocol = IPPROTO_UDP,
- };
- static const struct addrinfo prefbind = {
- .ai_flags = AI_PASSIVE,
- .ai_family = PF_INET,
- .ai_socktype = SOCK_DGRAM,
- .ai_protocol = IPPROTO_UDP,
- };
- static const int one = 1;
- int sndbuf, target_sndbuf = 131072;
- socklen_t len;
- char *sockname, *ssockname;
-
- res = get_address(&config->broadcast, &pref, &sockname);
- if(!res) exit(-1);
- if(config->broadcast_from.n) {
- sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
- if(!sres) exit(-1);
- } else
- sres = 0;
- if((bfd = socket(res->ai_family,
- res->ai_socktype,
- res->ai_protocol)) < 0)
- fatal(errno, "error creating broadcast socket");
- if(multicast(res->ai_addr)) {
- /* Multicasting */
- switch(res->ai_family) {
- case PF_INET: {
- const int mttl = config->multicast_ttl;
- if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0)
- fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
- if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_LOOP,
- &config->multicast_loop, sizeof one) < 0)
- fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
- break;
- }
- case PF_INET6: {
- const int mttl = config->multicast_ttl;
- if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
- &mttl, sizeof mttl) < 0)
- fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
- if(setsockopt(bfd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
- &config->multicast_loop, sizeof (int)) < 0)
- fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
- break;
- }
- default:
- fatal(0, "unsupported address family %d", res->ai_family);
- }
- info("multicasting on %s", sockname);
- } else {
- struct ifaddrs *ifs;
-
- if(getifaddrs(&ifs) < 0)
- fatal(errno, "error calling getifaddrs");
- while(ifs) {
- /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
- * still a null pointer. It turns out that there's a subsequent entry
- * for he same interface which _does_ have ifa_broadaddr though... */
- if((ifs->ifa_flags & IFF_BROADCAST)
- && ifs->ifa_broadaddr
- && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
- break;
- ifs = ifs->ifa_next;
- }
- if(ifs) {
- if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
- fatal(errno, "error setting SO_BROADCAST on broadcast socket");
- info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
- } else
- info("unicasting on %s", sockname);
- }
- len = sizeof sndbuf;
- if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
- &sndbuf, &len) < 0)
- fatal(errno, "error getting SO_SNDBUF");
- if(target_sndbuf > sndbuf) {
- if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
- &target_sndbuf, sizeof target_sndbuf) < 0)
- error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
- else
- info("changed socket send buffer size from %d to %d",
- sndbuf, target_sndbuf);
- } else
- info("default socket send buffer is %d",
- sndbuf);
- /* We might well want to set additional broadcast- or multicast-related
- * options here */
- if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
- fatal(errno, "error binding broadcast socket to %s", ssockname);
- if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error connecting broadcast socket to %s", sockname);
- /* Select an SSRC */
- gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
-}
-
-/** @brief Play over the network */
-static size_t network_play(size_t frames) {
- struct rtp_header header;
- struct iovec vec[2];
- size_t bytes = frames * bpf, written_frames;
- int written_bytes;
- /* We transmit using RTP (RFC3550) and attempt to conform to the internet
- * AVT profile (RFC3551). */
-
- /* If we're starting then initialize the base time */
- if(!rtp_time)
- xgettimeofday(&rtp_time_0, 0);
- if(idled) {
- /* There may have been a gap. Fix up the RTP time accordingly. */
- struct timeval now;
- uint64_t delta;
- uint64_t target_rtp_time;
-
- /* Find the current time */
- xgettimeofday(&now, 0);
- /* Find the number of microseconds elapsed since rtp_time=0 */
- delta = tvsub_us(now, rtp_time_0);
- if(delta > UINT64_MAX / 88200)
- fatal(0, "rtp_time=%"PRIu64" now=%ld.%06ld rtp_time_0=%ld.%06ld delta=%"PRIu64" (%"PRId64")",
- rtp_time,
- (long)now.tv_sec, (long)now.tv_usec,
- (long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
- delta, delta);
- target_rtp_time = (delta * config->sample_format.rate
- * config->sample_format.channels) / 1000000;
- /* Overflows at ~6 years uptime with 44100Hz stereo */
-
- /* rtp_time is the number of samples we've played. NB that we play
- * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
- * the value we deduce from time comparison.
- *
- * Suppose we have 1s track started at t=0, and another track begins to
- * play at t=2s. Suppose 44100Hz stereo. We send 1s of audio over the
- * next (about) one second, giving rtp_time=88200. rtp_time stops at this
- * point.
- *
- * At t=2s we'll have calculated target_rtp_time=176400. In this case we
- * set rtp_time=176400 and the player can correctly conclude that it
- * should leave 1s between the tracks.
- *
- * It's never right to reduce rtp_time, for that would imply packets with
- * overlapping timestamp ranges, which does not make sense.
- */
- target_rtp_time &= ~(uint64_t)1; /* stereo! */
- if(target_rtp_time > rtp_time) {
- /* More time has elapsed than we've transmitted samples. That implies
- * we've been 'sending' silence. */
- info("advancing rtp_time by %"PRIu64" samples",
- target_rtp_time - rtp_time);
- rtp_time = target_rtp_time;
- } else if(target_rtp_time < rtp_time) {
- info("would reverse rtp_time by %"PRIu64" samples",
- rtp_time - target_rtp_time);
- }
- }
- header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
- header.seq = htons(rtp_seq++);
- header.timestamp = htonl((uint32_t)rtp_time);
- header.ssrc = rtp_id;
- header.mpt = (idled ? 0x80 : 0x00) | 10;
- /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
- * the sample rate (in a library somewhere so that configuration.c can rule
- * out invalid rates).
- */
- idled = 0;
- if(bytes > NETWORK_BYTES - sizeof header) {
- bytes = NETWORK_BYTES - sizeof header;
- /* Always send a whole number of frames */
- bytes -= bytes % bpf;
- }
- /* "The RTP clock rate used for generating the RTP timestamp is independent
- * of the number of channels and the encoding; it equals the number of
- * sampling periods per second. For N-channel encodings, each sampling
- * period (say, 1/8000 of a second) generates N samples. (This terminology
- * is standard, but somewhat confusing, as the total number of samples
- * generated per second is then the sampling rate times the channel
- * count.)"
- */
- vec[0].iov_base = (void *)&header;
- vec[0].iov_len = sizeof header;
- vec[1].iov_base = playing->buffer + playing->start;
- vec[1].iov_len = bytes;
- do {
- written_bytes = writev(bfd, vec, 2);
- } while(written_bytes < 0 && errno == EINTR);
- if(written_bytes < 0) {
- error(errno, "error transmitting audio data");
- ++audio_errors;
- if(audio_errors == 10)
- fatal(0, "too many audio errors");
- return 0;
- } else
- audio_errors /= 2;
- written_bytes -= sizeof (struct rtp_header);
- written_frames = written_bytes / bpf;
- /* Advance RTP's notion of the time */
- rtp_time += written_frames * config->sample_format.channels;
- return written_frames;
-}
-
-static int bfd_slot;
-
-/** @brief Set up poll array for network play */
-static void network_beforepoll(int *timeoutp) {
- struct timeval now;
- uint64_t target_us;
- uint64_t target_rtp_time;
- const int64_t samples_per_second = config->sample_format.rate
- * config->sample_format.channels;
- int64_t lead, ahead_ms;
-
- /* If we're starting then initialize the base time */
- if(!rtp_time)
- xgettimeofday(&rtp_time_0, 0);
- /* We send audio data whenever we would otherwise get behind */
- xgettimeofday(&now, 0);
- target_us = tvsub_us(now, rtp_time_0);
- if(target_us > UINT64_MAX / 88200)
- fatal(0, "rtp_time=%"PRIu64" rtp_time_0=%ld.%06ld now=%ld.%06ld target_us=%"PRIu64" (%"PRId64")\n",
- rtp_time,
- (long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
- (long)now.tv_sec, (long)now.tv_usec,
- target_us, target_us);
- target_rtp_time = (target_us * config->sample_format.rate
- * config->sample_format.channels)
- / 1000000;
- /* Lead is how far ahead we are */
- lead = rtp_time - target_rtp_time;
- if(lead <= 0)
- /* We're behind or even, so we'll need to write as soon as we can */
- bfd_slot = addfd(bfd, POLLOUT);
- else {
- /* We've ahead, we can afford to wait a bit even if the IP stack thinks it
- * can accept more. */
- ahead_ms = 1000 * lead / samples_per_second;
- if(ahead_ms < *timeoutp)
- *timeoutp = ahead_ms;
- }
-}
-
-/** @brief Process poll() results for network play */
-static int network_ready(void) {
- if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
- return 1;
- else
- return 0;
-}
-
-const struct speaker_backend network_backend = {
- BACKEND_NETWORK,
- 0,
- network_init,
- 0, /* activate */
- network_play,
- 0, /* deactivate */
- network_beforepoll,
- network_ready
-};
-
-/*
-Local Variables:
-c-basic-offset:2
-comment-column:40
-fill-column:79
-indent-tabs-mode:nil
-End:
-*/
~mdw / disorder / blobdiff